[asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Kevin Larsen]

asterisk-users-bounces at lists.digium.com wrote on 10/28/2013 01:29:13 PM:

> From: Eddie Mikell <emikell at rimmkaufman.com>
> To: asterisk-users at lists.digium.com, 
> Date: 10/28/2013 01:29 PM
> Subject: [asterisk-users] Tired of dropouts and garbled phone calls 
> - where to go next?
> Sent by: asterisk-users-bounces at lists.digium.com
> 
> All,
> 
> The users in our organization are well, quite frankly, sick of phone
> service that is being provided.  The choppy phone calls, and drop 
> outs are detrimental to our sales force.
> 
> I've tried about everything I can think of.  
> 
> Moved the asterisk server from VM machine to dedicated machine
> More than enough bandwidth
> Setting 802.1p = 7
> Set Dedicated voice traffic 35% of bandwidth.
> 
> Not sure what option would be the best
> 
> Put analog lines in the conference room to avoid the dropouts - 
> leave the sip lines in place for day to day use
> Hire a consultant
> Ditch the system and buy a pre-packaged system - RingCentral or some 
such.
> 
> There are no local asterisk professionals who can help, and we are a
> little leery of opening up our system to outside consultants.
> 
> Anyone else face the above, and finally abandoned Asterisk for a 
> commercial system?  
> 
> We have 167 users.
> I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the
> conference rooms.
> 
> Suggestions welcome.
> 
> Best
> 
> Eddie

Does the garbled audio and dropouts only occur on outside calls, or do you 
get them on calls between extensions? How is your phone service delivered 
to your site?

If the extension to extension calls are clear, you need to be looking at 
your phone service and how you connect to it. If your local extensions are 
not clear, then you need to look at your asterisk implementation and your 
network.

If your incoming circuits are delivered via SIP, are you sure you have end 
to end QoS between your office and your phone provider? You can set all 
the QoS you want on the packets as they leave your network, but if your 
provider isn't honoring the QoS packets, then you can easily have audio 
issues. One of my locations has SIP service provided over a DSL line from 
the SIP provider. Just last week, the DSL line went down. We routed the 
calls out our standard internet connection and while it did work, we had 
audio dropouts (though only on incoming audio, the other end could hear us 
just fine). As soon as the DSL line was fixed and we routed back over 
their network, all the audio cleared up. It is the difference between low 
ping times and good QoS and higher ping times and providers who may not 
honor QoS.
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